Mixing system for mixing oversampled digital audio signals

ABSTRACT

A mixing system for mixing a plurality of digital audio signals, at least one of which is a noise-shaped oversampled digital audio signal having a predetermined sampling frequency and bit resolution, said system comprising a summing unit for computing a sum signal of said plurality of input signals; a clipping unit having an input for receiving said sum signal, said clipping unit clipping said sum signal; further comprising a filter unit between the input terminals and the clipping unit, arranged to selectively suppress frequency components outside an audio frequency band from the sum signal; and a converter unit arranged to receive a clipped sum signal from the clipping unit and to convert said clipped sum signal into an output signal of said bit resolution, using noise-shaping, the clipping unit being arranged to limit the input values to a range of values that the converter is able to handle in a stable manner.

The invention relates to a mixing system for mixing a plurality ofnoise-shaped oversampled digital audio signals having a predeterminedsampling frequency and bit resolution, said system comprising a summingunit having a plurality of input terminals each for receiving arespective one of said plurality of audio signals, for computing a sumsignal of said plurality of input signals, and a clipping unit having aninput for receiving said sum signal, said clipping unit clipping saidsum signal.

U.S. Pat. No. 5,581,480 discloses a mixer that sums a number of sampledaudio signals. The samples are represented using the well known PulseCode Modulation technology (PCM), wherein signal amplitudes arerepresented as multi-bit (for instance 8-16 bit) words. In thistechnology, a sampling frequency is used which is at least the minimalsampling frequency according to the Nyquist theorem, which, as is wellknown, is at least twice the signal bandwidth.

The mixer disclosed in this publication has a clipper for keeping themixed signal in a range of values that can be represented with the sameresolution as the input signals (generally 8 or 16-bit resolution). Sucha clipper reduces a sum of two audio samples to a lower value if saidsum exceeds a maximum value, for instance the maximum value that isrepresentable by 8 or 16 bits.

Although 8 or 16-bit PCM signals are a useful way of representing audiosignals, they have certain drawbacks due to their large word size. As analternative, it has been proposed to use “noise-shaped oversampled”audio signals instead of PCM signals with such a large word size.Noise-shaped oversampled audio signals involve one or a few bits persample at a sampling frequency well above the Nyquist frequency of theaudio signal. The basic idea of such signals is that the signal isrepresented in such a way that the spectral density of the largequantization errors that are the consequence of using a small number ofbits is concentrated at least substantially outside the audio bandwidthin the extra bandwidth available due the high sampling frequency.

As an example, in a format known as a standard DSD (Direct StreamDigital)-signal, audio contents are stored as a 1-bit sample stream witha sampling rate of 2.8 MHz. As an alternative, a slightly higherresolution, such as 2 bits per sample may be used.

Such noise-shaped oversampled sample streams offer a possibility to usea relatively low bit resolution with low audible noise, because inreconstructing an audio fragment from the samples, multiple samples maybe used to improve the signal resolution. Moreover, it has becomeapparent that the human auditory system appreciates this recordingtechnology better than the traditional PCM-recording technology, eventhough the sample stream, may have a very small, even one-bit,resolution.

It is desirable to mix such noise-shaped oversampled signals in such away that a mixed signal with the same sampling frequency is produced.Thus, a mixer can be incorporated in a system in which noise-shapedoversampled signals communicate between system components.

However, when such noise-shaped oversampled audio signals are mixed, themixed signal would be severely distorted if one uses clipping such asdisclosed in U.S. Pat. No. 5,581,480 to reduce the mixed signal to alower value if said sum exceeds a maximum value.

Therefore, it is an object of the invention to provide a mixing systemthat mixes noise-shaped oversampled signals to produce a noise-shapedoversampled output signal which suffers less from distortion.

To achieve the above-mentioned object, according to the invention, themixing system described in the preamble comprises:

-   -   a filter unit between the input terminals and the clipping unit,        arranged to selectively suppress frequency components outside an        audio frequency band in the input signals and/or the sum signal;        and    -   a converter unit arranged to receive a clipped sum signal from        the clipping unit and to convert said clipped sum signal into an        output signal of said bit resolution, using noise-shaping, the        clipping unit being arranged to limit the input values to a        range of values that the converter is able to handle in a stable        manner.

According to the invention, the converter converts the mixed signal backto the low resolution by means of noise shaping. The clipping unitlimits the input values to the converter, to a range that can be handledby the converter. This should be contrasted with the clipping unit ofU.S. Pat. No. 5,581,480, which functions to keep the signal amplitude inthe desired range representable by a specific PCM-bitword and therebyperforms the actual conversion function. The latter range is muchnarrower than the range that can be handled by the converter.

However, by applying a clipping operation, non-linear effects areintroduced. The non-linearity of said clipping function has the effectof folding back quantization noise from the input signals from above theaudio band back into the audible spectrum. In low-resolution signals,the quantization noise is relatively strong. The mixing system accordingto the invention suppresses high-frequency components from the signalbefore clipping, so as to reduce foldback due to the non-linearcharacter of said clipping operation.

In a preferred embodiment, said filter unit is comprised in an inputchannel and filters said input signals in order to limit an audiobandwidth of said input signals. Such a position has the advantage thatthe reduction of bandwidth mitigates the requirements on digitalprocessing speed. Preferably, said first and second sampling frequenciesare equal in magnitude; more specifically, said input signals and/orsaid output signals are of the above-mentioned DSD-format.

In a further preferred embodiment, said convertor unit comprises aSigma-Delta Modulator. Furthermore, the clipped signal may be maximizedto a clip level compliant with said Sigma-Delta Modulator. Specifically,said signal output may be maximized to −3 dB as compared to theamplitude output of the Sigma-Delta Modulator.

Said input channel may comprise a down-sampling unit for down-samplingsaid input signal. Such a down-sampling unit has the advantage that thereduction of bandwidth mitigates the requirements on digital processingspeed.

In order to output an output signal having the required second samplingfrequency, said convertor unit may comprise an up-sampling unit.

In order to achieve pleasant psycho-acoustical properties, the clippingunit may be of a soft clipping type.

The invention further relates to a method of mixing a plurality of audioinput signals having a first predetermined sampling frequency and bitresolution, said sampling frequency being relatively high with respectto an audio band width and said bit resolution being relatively low, themethod comprising the steps of receiving a respective one of saidplurality of audio signals, computing a sum signal of said plurality ofinput signals, selectively suppressing frequency components outside anaudio frequency band in the input signals and/or the sum signal,clipping said sum signal, and converting said clipped sum signal into anoutput signal of said bit resolution, using noise-shaping, the clippingunit being arranged to limit the input values to a range of values thatthe converter is able to handle in a stable manner.

The method may further comprise the step of limiting an audio bandwidthof said input signals.

The invention also relates to an audio system comprising a mixing systemaccording to the above-mentioned aspects, for mixing a plurality ofnoise-shaped oversampled digital audio signals having a predeterminedsampling frequency and bit resolution.

Further objects and features of the invention will become apparent fromthe drawings, wherein:

FIG. 1 is a schematic illustration of an audio system having a mixingsystem according to the invention.

FIG. 2 is a schematic illustration of an embodiment of the mixeraccording to the invention;

In FIG. 1, a generic setup of an audio playing system 1 is illustrated,which is able to utilize audio information contained in a multichannelrecording of a Super Audio CD 2 optimally for audio setups that havefewer output channels than a number of recorded channels. As an example,the number of input channels 3 in FIG. 2 is five, and the number ofoutput channels 4 is two. To this end, audio system 1 comprises a mixer5 for mixing said plurality of noise-shaped oversampled digital audiosignals 6 having a predetermined sampling frequency and bit resolution.An example of such a noise-shaped oversampled signal 6 is a DSD-signalformat, storing signal contents as a 1-bit sample stream with a (veryhigh) sampling rate of 2.8 MHz, in contrast to traditional recordingtechnologies (known as PCM or Pulse Code Modulation) which store signalsas multi-bit words. The format is able to provide an excellent audioquality and forms the standard for the successor to the conventional CDrecord carrier: the Super-Audio Compact Disc (SACD).

Although in this description, a sample stream is used, which isconstituted by single-bit samples, the sample stream may be formed inpractice by samples which are larger than one bit. The invention isapplicable in all cases where oversampling is applied in order toeliminate quantization noise effects due to a limited bit resolution ofthe bits used in the sample stream. After mixing, the setup of FIG. 1provides output signals 7 at the required sampling frequency and bitresolution, thereby creating the possibility to provide a modular signalprocessing system comprising processing modules 8 that are arranged toprovide a required trade-off in cost-effectiveness and/or quality.

In FIG. 2, mixer 5 comprises a plurality of input channels 3. In thisexample, each channel 3 comprises a down-sampling unit 9. By reducingthe sample rate of the input signals 6, more cost-effective signalprocessing is feasible by signal processors 10, present in said inputchannel 3.

After signal processing, the signal 6 is weighted with a scaling factorC1-CN in scaling stage 11. The signals 6 are then added in an addingunit 12, to produce a mixed signal 13. After adding, a clipping unit 14clips mixed signal 7 to limit said mixed signal 13 to a maximum signalamplitude.

Due to the clipping unit 14, non-linear effects are introduced in thesignal processing, which amount to frequency doubling and mixing.Therefore, frequency components related to quantization noise, which arepresent quite strongly in very high-frequency bands, are mapped backinto the audible spectrum. To eliminate such a foldback ofhigh-frequency noise, a filter unit 15 is introduced before clipper 14.This filter 15 is able to eliminate frequency components comprised insaid (mixed) signal 6, 13 originating from said bit resolution.

Filter 15 may be placed anywhere before the clipper to achieve thedesired filtering of high-frequency quantization noise. As an example,not depicted in FIG. 1, filter 15 may, in principle, be placed aftermixer 5. However, a preferable position of filter 15 is to combine sucha filter with down-sampling unit 9, as depicted in FIG. 1. In thisposition, a more cost-effective signal processing can be performed byreducing the reproduced audioband spectrum. When sufficiently reduced,the sampling frequency may also be reduced while maintaining anacceptable signal resolution at the same time.

In this way, the function of eliminating high-frequency quantizationnoise and reducing the reproduced audio spectrum are combined in asingle filter stage 15.

The output of adding unit 5 is multi-bit, due to various signalprocessing steps, scaling and adding of signal 6. To convert signal 6into the desired format of output signal 7, an up-sampling unit 16 and aconverter 17, preferably a Sigma-Delta Modulator are introduced. Such aconverter 17 is essentially a differentiator, outputting only incrementsof signal 13 as 1-bit values. Up-sampling may be achieved by a number ofwell-known routines, for example, sample-and-hold or interpolation.

Practical input values of a Sigma-Delta Modulator need to be under −3 dBin order to yield a stable outcome, so that the input voltage range islimited by −3 dB from the binary output voltages of the Sigma-DeltaModulator. The clipper 14 is preferably designed to limit the inputvalues inputted in converter 14.

In order to obtain a pleasant auditory perception of the outputtedsignal 7, a clipper 14 of the soft type is used, which limits the numberof higher order frequencies introduced by rounding of the edges of theclipping function.

It will be clear to those skilled in the art that the invention is notlimited to the embodiments described with reference to the drawing butmay comprise all kinds of variations thereof. These and other variationsare deemed to fall within the scope of protection of the appendedclaims.

1. A mixing system for mixing a plurality of digital audio signals, atleast one of which is a noise-shaped oversampled digital audio signalhaving a predetermined sampling frequency and bit resolution, saidsystem comprising: a summing unit having a plurality of input terminalseach for receiving a respective one of said plurality of digital audiosignals, for computing a sum signal of said plurality of digital audiosignals; a clipping unit having an input for receiving said sum signal,said clipping unit clipping said sum signal; a filter unit between theinput terminals and the clipping unit, arranged to selectively suppressfrequency components outside an audio frequency band from the sumsignal; and a converter unit arranged to receive a clipped sum signalfrom the clipping unit and to convert said clipped sum signal into anoutput signal of said bit resolution, using noise-shaping, the clippingunit being arranged to limit the input values to a range of values thatthe converter is able to handle in a stable manner.
 2. A mixing systemas claimed in claim 1, characterized in that said filter unit iscomprised in an input channel and filters said digital audio signals inorder to limit an audio bandwidth of said digital audio signals.
 3. Amixing system as claimed in claim 1, wherein sampling frequencies of thedigital audio signals and the output signal are equal in magnitude.
 4. Amixing system as claimed in claim 1, wherein said digital audio signalsand said output signal are of a DSD-format.
 5. A mixing system asclaimed in claim 1, characterized in that said convertor unit comprisesa Sigma-Delta Modulator.
 6. A mixing system as claimed in claim 5,characterized in that the clipped signal is maximized to a clip levelcompliant with said Sigma-Delta Modulator.
 7. A mixing system as claimedin claim 6, characterized in that said signal output is maximized to −3dB as compared to the amplitude output of the Sigma-Delta Modulator. 8.A mixing system as claimed in claim 1, further comprising one or moredown-sampling units for down-sampling said digital audio signals beforethe digital audio signals are applied to the summing unit.
 9. A mixingsystem as claimed in claim 1, wherein said convertor unit comprises anup-sampling unit.
 10. A mixing system as claimed in claim 1,characterized in that the clipping unit is of a soft clipping type. 11.An audio system comprising a mixing system as claimed in claim 1 formixing a plurality of noise-shaped oversampled digital audio signalshaving a predetermined sampling frequency and bit resolution.
 12. Themixing system of claim 2, wherein sampling frequencies of the digitalaudio signals and the output signal are equal in magnitude.
 13. Themixing system of claim 2, wherein said digital audio signals and saidoutput signal are of a DSD-format.
 14. The mixing system of claim 3,wherein said digital audio signals and said output signal are of aDSD-format.
 15. The mixing system of claim 2, wherein said convertorunit comprises a Sigma-Delta Modulator.
 16. The mixing system of claim3, wherein said convertor unit comprises a Sigma-Delta Modulator. 17.The mixing system of claim 4, wherein said convertor unit comprises aSigma-Delta Modulator.
 18. A method of mixing a plurality ofnoise-shaped oversampled digital audio signals having a predeterminedsampling frequency and bit resolution, the method comprising: receivinga respective one of said plurality of digital audio signals; computing asum signal of said plurality of digital audio signals; selectivelysuppressing frequency components outside an audio frequency band in atleast one of the digital audio signals and the sum signal; clipping saidsum signal; and converting said clipped sum signal into an output signalof said bit resolution, using noise-shaping, the clipping unit beingarranged to limit values of the sum signal to a range of values that theconverter is able to handle in a stable manner.
 19. A method as claimedin claim 18, wherein the method further comprises the step of limitingan audio bandwidth of said digital audio signals.
 20. A method asclaimed in claim 19, wherein the steps of selectively suppressingfrequency components outside an audio frequency band in at least one ofthe digital audio signals and the sum signal, and limiting an audiobandwidth of said input signals, are combined in a single stage.